Conventional circuit-switched telephony, known in the art as the public switched telephone network (PSTN), has the well-recognized disadvantage of requiring that a physical communications line be held open throughout an entire conversation between two communicating parties. A complex infrastructure of circuits and switches exists to establish and hold such lines for every PSTN telephone call. Because only a small percentage of the available bandwidth is actually used during each call, the PSTN system is inefficient and the infrastructure costs burdensome.
Newer, packet switching telephony technology integrates telephony with digital network technology. This packet switching technology codes telephony communications into digital packets, transfers the packets using non-dedicated, dynamically established routing between network servers, and decodes the digital packets at the receiving end to recreate the communication. Each packet may take a different path through a network, the connection between the parties being virtual, i.e. a permanent virtual connection (PVC), rather than physical in nature. Because packet switching telephony does not require a dedicated line, it is generally more efficient and cost-effective than PSTN telephony. However, packet switching communications requires complex software controls and communications commands, known as protocols, to establish, use and then close the virtual network connections between parties.
With the establishment of the Internet, the largest high-speed digital communications network in the world, protocols, such as voice over Internet protocols (VoIP), were established to support telephone communications over the Internet. One early set of protocols was known as H.323, a standard created by the International Telecommunications Union (ITU). A more recent set of VoIP protocols comprises Session Initiation Protocol (SIP) developed under the auspices of the Internet Engineering Task Force (IETF). Both H.323 and SIP provide standardized protocols for VoIP telephony, with SIP considered to be somewhat simpler and more flexible than H.323.
As VoIP telephony has become more prevalent, it has concomitantly become necessary for voice to share hardware and software resources in packet switching networks with other types of data. As expected, each of the different data types has it's own service requirements. Voice service, for example, requires relatively low throughput, low delay and low packet loss. In contrast, file transfer protocol (FTP) data, operating under TCP protocols, requires relatively higher throughput but will tolerate longer delays and potentially higher packet loss. For each different service type, service requirements define the minimally acceptable network behavior for that service type. QoS parameters are derived for an Internet protocol (IP) network to meet those service requirements. As will be described in further detail below, QoS can be measured on different networks for different data types. Network operations can be adjusted to meet the QoS requirements for various data types.
For VoIP telephony, voice service standards were established by the ITU and published as ITU-T G114, the standards defined with respect to a variety of parameters including: one-way delay, jitter, throughput (dependent on coding algorithms employed), coding cycle restrictions, packet loss and service availability. In a wide area network (WAN) environment such as a typical enterprise WAN, a permanent virtual circuit is established between the transmission and receiver sources. Enterprise WANs consist of a variety of hardware components, the principal components comprising routers and switches, which are commercially available, programmable network nodes; they are capable of receiving and transmitting data packets, including VoIP data packets, in accordance with appropriate protocols. Routers and switches are programmable by the operator, each of them generally being set up in a manner to optimally handle anticipated network traffic.
Network routers inherently develop queues of data packets, the data packets processed in accordance with the programmed, controllable settings of the routers. These router settings, along with the resultant handling of the data packets, directly affect the service levels of the various data services handled by the routers. Routers are thus provided with adjustable operating parameters, the operating parameters typically set by software. Adjustable router operating parameters typically include parameters that affect QoS and thus directly affect voice and data service levels. Router QoS ‘tool chests’ enable operators to set these QoS parameters to control the interaction of voice and data in the network. However, adjusting router settings to meet QoS requirements is a complex effort. As noted above, different types of data services have different service requirements. As voice becomes integrated with data through VoIP telephony, QoS parameters must be set so that the service requirements for telephony are met in addition to the service requirements for other data types. Further, many geographically disperse routers handling many different data applications are required to support VoIP, making the configuration of those routers complex and difficult.
Different conceptualizations of and at least partial solutions are known to these problems in the art. For example, U.S. Pat. No. 6,430,154 to Hunt et al. shows methods and systems for supporting multiple application traffic types on networks where elastic and inelastic data are determined and handled differently. U.S. Pat. No. 6,510,219 to Wellard et al. shows an alternate network fallback solution for IP telephony wherein QoS is monitored and, if it falls below a first threshold, a connection over an alternate network is established. If the QoS falls below another predetermined threshold, the call is transferred to the alternate network connection. U.S. Pat. No. 6,515,963 to Bechtolsheim et al. shows a per-flow dynamic buffer management scheme for a data communications device.
There exists in the art a need for managing the service level of VoIP telephony in WANs operating multiple routers distributed over geographically disperse areas and supporting multiple data applications.